Files
ppsspp/Core/HW/SasAudio.cpp
T
Unknown W. Brackets 6b413a808f Atrac: Handle SAS end the same way as other types.
It makes sense to match, and this sounds right in Sol Trigger.  Haven't
tested in detail yet, but it should prevent cutting off sounds.
2015-11-08 12:23:32 -08:00

954 lines
24 KiB
C++

// Copyright (c) 2012- PPSSPP Project.
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, version 2.0 or later versions.
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License 2.0 for more details.
// A copy of the GPL 2.0 should have been included with the program.
// If not, see http://www.gnu.org/licenses/
// Official git repository and contact information can be found at
// https://github.com/hrydgard/ppsspp and http://www.ppsspp.org/.
#include "base/basictypes.h"
#include "profiler/profiler.h"
#include "Globals.h"
#include "Core/MemMapHelpers.h"
#include "Core/HLE/sceAtrac.h"
#include "Core/Config.h"
#include "Core/Reporting.h"
#include "SasAudio.h"
#include <algorithm>
// #define AUDIO_TO_FILE
static const u8 f[16][2] = {
{ 0, 0 },
{ 60, 0 },
{ 115, 52 },
{ 98, 55 },
{ 122, 60 },
// TODO: The below values could use more testing, but match initial tests.
// Not sure if they are used by games, found by tests.
{ 0, 0 },
{ 0, 0 },
{ 52, 0 },
{ 55, 2 },
{ 60, 125 },
{ 0, 0 },
{ 0, 91 },
{ 0, 0 },
{ 2, 216 },
{ 125, 6 },
{ 0, 151 },
};
void VagDecoder::Start(u32 data, u32 vagSize, bool loopEnabled) {
loopEnabled_ = loopEnabled;
loopAtNextBlock_ = false;
loopStartBlock_ = -1;
numBlocks_ = vagSize / 16;
end_ = false;
data_ = data;
read_ = data;
curSample = 28;
curBlock_ = -1;
s_1 = 0; // per block?
s_2 = 0;
}
void VagDecoder::DecodeBlock(u8 *&read_pointer) {
u8 *readp = read_pointer;
int predict_nr = *readp++;
int shift_factor = predict_nr & 0xf;
predict_nr >>= 4;
int flags = *readp++;
if (flags == 7) {
VERBOSE_LOG(SASMIX, "VAG ending block at %d", curBlock_);
end_ = true;
return;
}
else if (flags == 6) {
loopStartBlock_ = curBlock_;
}
else if (flags == 3) {
if (loopEnabled_) {
loopAtNextBlock_ = true;
}
}
// Keep state in locals to avoid bouncing to memory.
int s1 = s_1;
int s2 = s_2;
int coef1 = f[predict_nr][0];
int coef2 = -f[predict_nr][1];
// TODO: Unroll once more and interleave the unpacking with the decoding more?
for (int i = 0; i < 28; i += 2) {
u8 d = *readp++;
int sample1 = (short)((d & 0xf) << 12) >> shift_factor;
int sample2 = (short)((d & 0xf0) << 8) >> shift_factor;
s2 = clamp_s16(sample1 + ((s1 * coef1 + s2 * coef2) >> 6));
s1 = clamp_s16(sample2 + ((s2 * coef1 + s1 * coef2) >> 6));
samples[i] = s2;
samples[i + 1] = s1;
}
s_1 = s1;
s_2 = s2;
curSample = 0;
curBlock_++;
if (curBlock_ == numBlocks_) {
end_ = true;
}
read_pointer = readp;
}
void VagDecoder::GetSamples(s16 *outSamples, int numSamples) {
if (end_) {
memset(outSamples, 0, numSamples * sizeof(s16));
return;
}
if (!Memory::IsValidAddress(read_)) {
WARN_LOG(SASMIX, "Bad VAG samples address?");
return;
}
u8 *readp = Memory::GetPointerUnchecked(read_);
u8 *origp = readp;
for (int i = 0; i < numSamples; i++) {
if (curSample == 28) {
if (loopAtNextBlock_) {
VERBOSE_LOG(SASMIX, "Looping VAG from block %d/%d to %d", curBlock_, numBlocks_, loopStartBlock_);
// data_ starts at curBlock = -1.
read_ = data_ + 16 * loopStartBlock_ + 16;
readp = Memory::GetPointerUnchecked(read_);
origp = readp;
curBlock_ = loopStartBlock_;
loopAtNextBlock_ = false;
}
DecodeBlock(readp);
if (end_) {
// Clear the rest of the buffer and return.
memset(&outSamples[i], 0, (numSamples - i) * sizeof(s16));
return;
}
}
outSamples[i] = samples[curSample++];
}
if (readp > origp) {
read_ += readp - origp;
}
}
void VagDecoder::DoState(PointerWrap &p) {
auto s = p.Section("VagDecoder", 1, 2);
if (!s)
return;
if (s >= 2) {
p.DoArray(samples, ARRAY_SIZE(samples));
} else {
int samplesOld[ARRAY_SIZE(samples)];
p.DoArray(samplesOld, ARRAY_SIZE(samples));
for (size_t i = 0; i < ARRAY_SIZE(samples); ++i) {
samples[i] = samplesOld[i];
}
}
p.Do(curSample);
p.Do(data_);
p.Do(read_);
p.Do(curBlock_);
p.Do(loopStartBlock_);
p.Do(numBlocks_);
p.Do(s_1);
p.Do(s_2);
p.Do(loopEnabled_);
p.Do(loopAtNextBlock_);
p.Do(end_);
}
int SasAtrac3::setContext(u32 context) {
contextAddr_ = context;
atracID_ = _AtracGetIDByContext(context);
if (!sampleQueue_)
sampleQueue_ = new BufferQueue();
sampleQueue_->clear();
end_ = false;
return 0;
}
void SasAtrac3::getNextSamples(s16 *outbuf, int wantedSamples) {
if (atracID_ < 0) {
end_ = true;
return;
}
u32 finish = 0;
int wantedbytes = wantedSamples * sizeof(s16);
while (!finish && sampleQueue_->getQueueSize() < wantedbytes) {
u32 numSamples = 0;
int remains = 0;
static s16 buf[0x800];
_AtracDecodeData(atracID_, (u8*)buf, 0, &numSamples, &finish, &remains);
if (numSamples > 0)
sampleQueue_->push((u8*)buf, numSamples * sizeof(s16));
else
finish = 1;
}
sampleQueue_->pop_front((u8*)outbuf, wantedbytes);
end_ = finish == 1;
}
int SasAtrac3::addStreamData(u32 bufPtr, u32 addbytes) {
if (atracID_ > 0) {
_AtracAddStreamData(atracID_, bufPtr, addbytes);
}
return 0;
}
void SasAtrac3::DoState(PointerWrap &p) {
auto s = p.Section("SasAtrac3", 1, 2);
if (!s)
return;
p.Do(contextAddr_);
p.Do(atracID_);
if (p.mode == p.MODE_READ && atracID_ >= 0 && !sampleQueue_) {
sampleQueue_ = new BufferQueue();
}
if (s >= 2) {
p.Do(end_);
}
}
// http://code.google.com/p/jpcsp/source/browse/trunk/src/jpcsp/HLE/modules150/sceSasCore.java
static int simpleRate(int n) {
n &= 0x7F;
if (n == 0x7F) {
return 0;
}
int rate = ((7 - (n & 0x3)) << 26) >> (n >> 2);
if (rate == 0) {
return 1;
}
return rate;
}
static int exponentRate(int n) {
n &= 0x7F;
if (n == 0x7F) {
return 0;
}
int rate = ((7 - (n & 0x3)) << 24) >> (n >> 2);
if (rate == 0) {
return 1;
}
return rate;
}
static int getAttackRate(int bitfield1) {
return simpleRate(bitfield1 >> 8);
}
static int getAttackType(int bitfield1) {
return (bitfield1 & 0x8000) == 0 ? PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE : PSP_SAS_ADSR_CURVE_MODE_LINEAR_BENT;
}
static int getDecayRate(int bitfield1) {
int n = (bitfield1 >> 4) & 0x000F;
if (n == 0)
return 0x7FFFFFFF;
return 0x80000000 >> n;
}
static int getSustainType(int bitfield2) {
return (bitfield2 >> 14) & 3;
}
static int getSustainRate(int bitfield2) {
if (getSustainType(bitfield2) == PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE) {
return exponentRate(bitfield2 >> 6);
} else {
return simpleRate(bitfield2 >> 6);
}
}
static int getReleaseType(int bitfield2) {
return (bitfield2 & 0x0020) == 0 ? PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE : PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE;
}
static int getReleaseRate(int bitfield2) {
int n = bitfield2 & 0x001F;
if (n == 31) {
return 0;
}
if (getReleaseType(bitfield2) == PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE) {
if (n == 30) {
return 0x40000000;
} else if (n == 29) {
return 1;
}
return 0x10000000 >> n;
}
if (n == 0)
return 0x7FFFFFFF;
return 0x80000000 >> n;
}
static int getSustainLevel(int bitfield1) {
return ((bitfield1 & 0x000F) + 1) << 26;
}
void ADSREnvelope::SetSimpleEnvelope(u32 ADSREnv1, u32 ADSREnv2) {
attackRate = getAttackRate(ADSREnv1);
attackType = getAttackType(ADSREnv1);
decayRate = getDecayRate(ADSREnv1);
decayType = PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE;
sustainRate = getSustainRate(ADSREnv2);
sustainType = getSustainType(ADSREnv2);
releaseRate = getReleaseRate(ADSREnv2);
releaseType = getReleaseType(ADSREnv2);
sustainLevel = getSustainLevel(ADSREnv1);
if (attackRate < 0 || decayRate < 0 || sustainRate < 0 || releaseRate < 0) {
ERROR_LOG_REPORT(SCESAS, "Simple ADSR resulted in invalid rates: %04x, %04x", ADSREnv1, ADSREnv2);
}
}
SasInstance::SasInstance()
: maxVoices(PSP_SAS_VOICES_MAX),
sampleRate(44100),
outputMode(PSP_SAS_OUTPUTMODE_MIXED),
mixBuffer(0),
sendBuffer(0),
sendBufferDownsampled(0),
sendBufferProcessed(0),
resampleBuffer(0),
grainSize(0) {
#ifdef AUDIO_TO_FILE
audioDump = fopen("D:\\audio.raw", "wb");
#endif
memset(&waveformEffect, 0, sizeof(waveformEffect));
waveformEffect.type = PSP_SAS_EFFECT_TYPE_OFF;
waveformEffect.isDryOn = 1;
}
SasInstance::~SasInstance() {
ClearGrainSize();
}
void SasInstance::GetDebugText(char *text, size_t bufsize) {
char voiceBuf[4096];
voiceBuf[0] = '\0';
char *p = voiceBuf;
for (int i = 0; i < maxVoices; i++) {
if (voices[i].playing) {
p += snprintf(p, sizeof(voiceBuf) - (p - voiceBuf), " %d: Pitch %d L/R,FX: %d,%d|%d,%d VAG: %08x:%d:%08x Height:%d%%\n", i, voices[i].pitch, voices[i].volumeLeft, voices[i].volumeRight, voices[i].effectLeft, voices[i].effectRight, voices[i].vagAddr, voices[i].vagSize, voices[i].vag.GetReadPtr(), (int)((int64_t)voices[i].envelope.GetHeight() * 100 / PSP_SAS_ENVELOPE_HEIGHT_MAX));
}
}
snprintf(text, bufsize,
"SR: %d Mode: %s Grain: %d\n"
"Effect: Type: %d Dry: %d Wet: %d L: %d R: %d Delay: %d Feedback: %d\n"
"\n%s\n",
sampleRate, outputMode == PSP_SAS_OUTPUTMODE_RAW ? "Raw" : "Mixed", grainSize,
waveformEffect.type, waveformEffect.isDryOn, waveformEffect.isWetOn, waveformEffect.leftVol, waveformEffect.rightVol, waveformEffect.delay, waveformEffect.feedback,
voiceBuf);
}
void SasInstance::ClearGrainSize() {
delete[] mixBuffer;
delete[] sendBuffer;
delete[] sendBufferDownsampled;
delete[] sendBufferProcessed;
delete[] resampleBuffer;
mixBuffer = nullptr;
sendBuffer = nullptr;
resampleBuffer = nullptr;
sendBufferDownsampled = nullptr;
sendBufferProcessed = nullptr;
}
void SasInstance::SetGrainSize(int newGrainSize) {
grainSize = newGrainSize;
// If you change the sizes here, don't forget DoState().
delete[] mixBuffer;
delete[] sendBuffer;
delete[] sendBufferDownsampled;
delete[] sendBufferProcessed;
delete[] resampleBuffer;
mixBuffer = new s32[grainSize * 2];
sendBuffer = new s32[grainSize * 2];
sendBufferDownsampled = new s16[grainSize];
sendBufferProcessed = new s16[grainSize * 2];
memset(mixBuffer, 0, sizeof(int) * grainSize * 2);
memset(sendBuffer, 0, sizeof(int) * grainSize * 2);
memset(sendBufferDownsampled, 0, sizeof(s16) * grainSize);
memset(sendBufferProcessed, 0, sizeof(s16) * grainSize * 2);
// 2 samples padding at the start, that's where we copy the two last samples from the channel
// so that we can do bicubic resampling if necessary. Plus 1 for smoothness hackery.
resampleBuffer = new s16[grainSize * 4 + 3];
}
void SasVoice::ReadSamples(s16 *output, int numSamples) {
// Read N samples into the resample buffer. Could do either PCM or VAG here.
switch (type) {
case VOICETYPE_VAG:
vag.GetSamples(output, numSamples);
break;
case VOICETYPE_PCM:
{
int needed = numSamples;
s16 *out = output;
while (needed > 0) {
u32 size = std::min(pcmSize - pcmIndex, needed);
if (!on) {
pcmIndex = 0;
break;
}
Memory::Memcpy(out, pcmAddr + pcmIndex * sizeof(s16), size * sizeof(s16));
pcmIndex += size;
needed -= size;
out += size;
if (pcmIndex >= pcmSize) {
if (!loop) {
// All out, quit. We'll end in HaveSamplesEnded().
break;
}
pcmIndex = pcmLoopPos;
}
}
if (needed > 0) {
memset(out, 0, needed * sizeof(s16));
}
}
break;
case VOICETYPE_ATRAC3:
atrac3.getNextSamples(output, numSamples);
break;
default:
{
memset(output, 0, numSamples * sizeof(s16));
}
break;
}
}
bool SasVoice::HaveSamplesEnded() const {
switch (type) {
case VOICETYPE_VAG:
return vag.End();
case VOICETYPE_PCM:
return pcmIndex >= pcmSize;
case VOICETYPE_ATRAC3:
return atrac3.End();
default:
return false;
}
}
void SasInstance::MixVoice(SasVoice &voice) {
switch (voice.type) {
case VOICETYPE_VAG:
if (voice.type == VOICETYPE_VAG && !voice.vagAddr)
break;
// else fallthrough! Don't change the check above.
case VOICETYPE_PCM:
if (voice.type == VOICETYPE_PCM && !voice.pcmAddr)
break;
// else fallthrough! Don't change the check above.
default:
// Load resample history (so we can use a wide filter)
resampleBuffer[0] = voice.resampleHist[0];
resampleBuffer[1] = voice.resampleHist[1];
// Figure out number of samples to read.
// Actually this is not entirely correct - we need to get one extra sample, and store it
// for the next time around. A little complicated...
// But for now, see Smoothness HACKERY below :P
u32 numSamples = ((u32)voice.sampleFrac + (u32)grainSize * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
if ((int)numSamples > grainSize * 4) {
ERROR_LOG(SASMIX, "numSamples too large, clamping: %i vs %i", numSamples, grainSize * 4);
numSamples = grainSize * 4;
}
// This feels a bit hacky. The first 32 samples after a keyon are 0s.
const bool ignorePitch = voice.type == VOICETYPE_PCM && voice.pitch > PSP_SAS_PITCH_BASE;
if (voice.envelope.NeedsKeyOn()) {
int delay = ignorePitch ? 32 : (32 * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
// VAG seems to have an extra sample delay (not shared by PCM.)
if (voice.type == VOICETYPE_VAG)
++delay;
voice.ReadSamples(resampleBuffer + 2 + delay, numSamples - delay);
} else {
voice.ReadSamples(resampleBuffer + 2, numSamples);
}
// Smoothness HACKERY
resampleBuffer[2 + numSamples] = resampleBuffer[2 + numSamples - 1];
// Save resample history
voice.resampleHist[0] = resampleBuffer[2 + numSamples - 2];
voice.resampleHist[1] = resampleBuffer[2 + numSamples - 1];
// Resample to the correct pitch, writing exactly "grainSize" samples.
// This is a HORRIBLE resampler by the way.
// TODO: Special case no-resample case (and 2x and 0.5x) for speed, it's not uncommon
u32 sampleFrac = voice.sampleFrac;
for (int i = 0; i < grainSize; i++) {
// For now: nearest neighbour, not even using the resample history at all.
int sample = resampleBuffer[sampleFrac / PSP_SAS_PITCH_BASE + 2];
sampleFrac += voice.pitch;
// The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1.
// Reduce it to 14 bits, by shifting off 15. Round up by adding (1 << 14) first.
int envelopeValue = voice.envelope.GetHeight();
voice.envelope.Step();
envelopeValue = (envelopeValue + (1 << 14)) >> 15;
// We just scale by the envelope before we scale by volumes.
// Again, we round up by adding (1 << 14) first (*after* multiplying.)
sample = ((sample * envelopeValue) + (1 << 14)) >> 15;
// We mix into this 32-bit temp buffer and clip in a second loop
// Ideally, the shift right should be there too but for now I'm concerned about
// not overflowing.
mixBuffer[i * 2] += (sample * voice.volumeLeft ) >> 12;
mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> 12;
sendBuffer[i * 2] += sample * voice.effectLeft >> 12;
sendBuffer[i * 2 + 1] += sample * voice.effectRight >> 12;
}
voice.sampleFrac = sampleFrac;
// Let's hope grainSize is a power of 2.
//voice.sampleFrac &= grainSize * PSP_SAS_PITCH_BASE - 1;
voice.sampleFrac -= numSamples * PSP_SAS_PITCH_BASE;
if (voice.HaveSamplesEnded())
voice.envelope.End();
if (voice.envelope.HasEnded())
{
// NOTICE_LOG(SCESAS, "Hit end of envelope");
voice.playing = false;
voice.on = false;
}
}
}
void SasInstance::Mix(u32 outAddr, u32 inAddr, int leftVol, int rightVol) {
PROFILE_THIS_SCOPE("mixer");
int voicesPlayingCount = 0;
for (int v = 0; v < PSP_SAS_VOICES_MAX; v++) {
SasVoice &voice = voices[v];
if (!voice.playing || voice.paused)
continue;
voicesPlayingCount++;
MixVoice(voice);
}
// Then mix the send buffer in with the rest.
// Alright, all voices mixed. Let's convert and clip, and at the same time, wipe mixBuffer for next time. Could also dither.
s16 *outp = (s16 *)Memory::GetPointer(outAddr);
const s16 *inp = inAddr ? (s16*)Memory::GetPointer(inAddr) : 0;
if (outputMode == PSP_SAS_OUTPUTMODE_MIXED) {
// Okay, apply effects processing to the Send buffer.
WriteMixedOutput(outp, inp, leftVol, rightVol);
} else {
s16 *outpL = outp + grainSize * 0;
s16 *outpR = outp + grainSize * 1;
s16 *outpSendL = outp + grainSize * 2;
s16 *outpSendR = outp + grainSize * 3;
WARN_LOG_REPORT_ONCE(sasraw, SCESAS, "sceSasCore: raw outputMode");
for (int i = 0; i < grainSize * 2; i += 2) {
*outpL++ = clamp_s16(mixBuffer[i + 0]);
*outpR++ = clamp_s16(mixBuffer[i + 1]);
*outpSendL++ = clamp_s16(sendBuffer[i + 0]);
*outpSendR++ = clamp_s16(sendBuffer[i + 1]);
}
}
memset(mixBuffer, 0, grainSize * sizeof(int) * 2);
memset(sendBuffer, 0, grainSize * sizeof(int) * 2);
#ifdef AUDIO_TO_FILE
fwrite(Memory::GetPointer(outAddr), 1, grainSize * 2 * 2, audioDump);
#endif
}
void SasInstance::WriteMixedOutput(s16 *outp, const s16 *inp, int leftVol, int rightVol) {
const bool dry = waveformEffect.isDryOn != 0;
const bool wet = waveformEffect.isWetOn != 0;
if (wet) {
ApplyWaveformEffect();
}
if (inp) {
for (int i = 0; i < grainSize * 2; i += 2) {
int sampleL = ((*inp++) * leftVol >> 12);
int sampleR = ((*inp++) * rightVol >> 12);
if (dry) {
sampleL += mixBuffer[i + 0];
sampleR += mixBuffer[i + 1];
}
if (wet) {
sampleL += sendBufferProcessed[i + 0];
sampleR += sendBufferProcessed[i + 1];
}
*outp++ = clamp_s16(sampleL);
*outp++ = clamp_s16(sampleR);
}
} else {
// These are the optimal cases.
if (dry && wet) {
for (int i = 0; i < grainSize * 2; i += 2) {
*outp++ = clamp_s16(mixBuffer[i + 0] + sendBufferProcessed[i + 0]);
*outp++ = clamp_s16(mixBuffer[i + 1] + sendBufferProcessed[i + 1]);
}
} else if (dry) {
for (int i = 0; i < grainSize * 2; i += 2) {
*outp++ = clamp_s16(mixBuffer[i + 0]);
*outp++ = clamp_s16(mixBuffer[i + 1]);
}
} else {
// This is another uncommon case, dry must be off but let's keep it for clarity.
for (int i = 0; i < grainSize * 2; i += 2) {
int sampleL = 0;
int sampleR = 0;
if (dry) {
sampleL += mixBuffer[i + 0];
sampleR += mixBuffer[i + 1];
}
if (wet) {
sampleL += sendBufferProcessed[i + 0];
sampleR += sendBufferProcessed[i + 1];
}
*outp++ = clamp_s16(sampleL);
*outp++ = clamp_s16(sampleR);
}
}
}
}
void SasInstance::SetWaveformEffectType(int type) {
if (type != waveformEffect.type) {
waveformEffect.type = type;
reverb_.SetPreset(type);
}
}
// http://psx.rules.org/spu.txt has some information about setting up the delay time by modifying the delay preset.
// See http://report.ppsspp.org/logs/kind/772 for a list of games that use different types. Maybe can help us figure out
// which is which.
void SasInstance::ApplyWaveformEffect() {
// First, downsample the send buffer to 22khz. We do this naively for now.
for (int i = 0; i < grainSize / 2; i++) {
sendBufferDownsampled[i * 2] = clamp_s16(sendBuffer[i * 4]);
sendBufferDownsampled[i * 2 + 1] = clamp_s16(sendBuffer[i * 4 + 1]);
}
// Volume max is 0x1000, while our factor is up to 0x8000. Shifting right by 3 fixes that.
reverb_.ProcessReverb(sendBufferProcessed, sendBufferDownsampled, grainSize / 2, waveformEffect.leftVol << 3, waveformEffect.rightVol << 3);
}
void SasInstance::DoState(PointerWrap &p) {
auto s = p.Section("SasInstance", 1);
if (!s)
return;
p.Do(grainSize);
if (p.mode == p.MODE_READ) {
if (grainSize > 0) {
SetGrainSize(grainSize);
} else {
ClearGrainSize();
}
}
p.Do(maxVoices);
p.Do(sampleRate);
p.Do(outputMode);
// SetGrainSize() / ClearGrainSize() should've made our buffers match.
if (mixBuffer != NULL && grainSize > 0) {
p.DoArray(mixBuffer, grainSize * 2);
}
if (sendBuffer != NULL && grainSize > 0) {
p.DoArray(sendBuffer, grainSize * 2);
}
if (resampleBuffer != NULL && grainSize > 0) {
p.DoArray(resampleBuffer, grainSize * 4 + 3);
}
int n = PSP_SAS_VOICES_MAX;
p.Do(n);
if (n != PSP_SAS_VOICES_MAX) {
ERROR_LOG(HLE, "Savestate failure: wrong number of SAS voices");
return;
}
p.DoArray(voices, ARRAY_SIZE(voices));
p.Do(waveformEffect);
}
void SasVoice::Reset() {
resampleHist[0] = 0;
resampleHist[1] = 0;
}
void SasVoice::KeyOn() {
envelope.KeyOn();
switch (type) {
case VOICETYPE_VAG:
if (Memory::IsValidAddress(vagAddr)) {
vag.Start(vagAddr, vagSize, loop);
} else {
ERROR_LOG(SASMIX, "Invalid VAG address %08x", vagAddr);
return;
}
break;
default:
break;
}
playing = true;
on = true;
paused = false;
sampleFrac = 0;
}
void SasVoice::KeyOff() {
on = false;
envelope.KeyOff();
}
void SasVoice::ChangedParams(bool changedVag) {
if (!playing && on) {
playing = true;
if (changedVag)
vag.Start(vagAddr, vagSize, loop);
}
// TODO: restart VAG somehow
}
void SasVoice::DoState(PointerWrap &p)
{
auto s = p.Section("SasVoice", 1, 3);
if (!s)
return;
p.Do(playing);
p.Do(paused);
p.Do(on);
p.Do(type);
p.Do(vagAddr);
p.Do(vagSize);
p.Do(pcmAddr);
p.Do(pcmSize);
p.Do(pcmIndex);
if (s >= 2) {
p.Do(pcmLoopPos);
} else {
pcmLoopPos = 0;
}
p.Do(sampleRate);
p.Do(sampleFrac);
p.Do(pitch);
p.Do(loop);
if (s < 2 && type == VOICETYPE_PCM) {
// We set loop incorrectly before, and always looped.
// Let's keep always looping, since it's usually right.
loop = true;
}
p.Do(noiseFreq);
p.Do(volumeLeft);
p.Do(volumeRight);
if (s < 3) {
// There were extra variables here that were for the same purpose.
p.Do(effectLeft);
p.Do(effectRight);
}
p.Do(effectLeft);
p.Do(effectRight);
p.DoArray(resampleHist, ARRAY_SIZE(resampleHist));
envelope.DoState(p);
vag.DoState(p);
atrac3.DoState(p);
}
ADSREnvelope::ADSREnvelope()
: attackRate(0),
decayRate(0),
sustainRate(0),
releaseRate(0),
attackType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE),
decayType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE),
sustainType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE),
sustainLevel(0),
releaseType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE),
state_(STATE_OFF),
height_(0) {
}
void ADSREnvelope::WalkCurve(int type, int rate) {
s64 expDelta;
switch (type) {
case PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE:
height_ += rate;
break;
case PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE:
height_ -= rate;
break;
case PSP_SAS_ADSR_CURVE_MODE_LINEAR_BENT:
if (height_ <= (s64)PSP_SAS_ENVELOPE_HEIGHT_MAX * 3 / 4) {
height_ += rate;
} else {
height_ += rate / 4;
}
break;
case PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE:
expDelta = height_ - PSP_SAS_ENVELOPE_HEIGHT_MAX;
// Flipping the sign so that we can shift in the top bits.
expDelta += (-expDelta * rate) >> 32;
height_ = expDelta + PSP_SAS_ENVELOPE_HEIGHT_MAX - (rate + 3UL) / 4UL;
break;
case PSP_SAS_ADSR_CURVE_MODE_EXPONENT_INCREASE:
expDelta = height_ - PSP_SAS_ENVELOPE_HEIGHT_MAX;
// Flipping the sign so that we can shift in the top bits.
expDelta += (-expDelta * rate) >> 32;
height_ = expDelta + 0x4000 + PSP_SAS_ENVELOPE_HEIGHT_MAX;
break;
case PSP_SAS_ADSR_CURVE_MODE_DIRECT:
height_ = rate; // Simple :)
break;
}
}
void ADSREnvelope::SetState(ADSRState state) {
if (height_ > PSP_SAS_ENVELOPE_HEIGHT_MAX) {
height_ = PSP_SAS_ENVELOPE_HEIGHT_MAX;
}
// TODO: Also check for height_ < 0 and set to 0?
state_ = state;
}
inline void ADSREnvelope::Step() {
switch (state_) {
case STATE_ATTACK:
WalkCurve(attackType, attackRate);
if (height_ >= PSP_SAS_ENVELOPE_HEIGHT_MAX || height_ < 0)
SetState(STATE_DECAY);
break;
case STATE_DECAY:
WalkCurve(decayType, decayRate);
if (height_ < sustainLevel)
SetState(STATE_SUSTAIN);
break;
case STATE_SUSTAIN:
WalkCurve(sustainType, sustainRate);
if (height_ <= 0) {
height_ = 0;
SetState(STATE_RELEASE);
}
break;
case STATE_RELEASE:
WalkCurve(releaseType, releaseRate);
if (height_ <= 0) {
height_ = 0;
SetState(STATE_OFF);
}
break;
case STATE_OFF:
// Do nothing
break;
case STATE_KEYON:
height_ = 0;
SetState(STATE_KEYON_STEP);
break;
case STATE_KEYON_STEP:
// This entire state is pretty much a hack to reproduce PSP behavior.
// The STATE_KEYON state is a real state, but not sure how it switches.
// It takes 32 steps at 0 for keyon to "kick in", 31 should shift to 0 anyway.
height_++;
if (height_ >= 31) {
height_ = 0;
SetState(STATE_ATTACK);
}
break;
}
}
void ADSREnvelope::KeyOn() {
SetState(STATE_KEYON);
}
void ADSREnvelope::KeyOff() {
SetState(STATE_RELEASE);
}
void ADSREnvelope::End() {
SetState(STATE_OFF);
height_ = 0;
}
void ADSREnvelope::DoState(PointerWrap &p) {
auto s = p.Section("ADSREnvelope", 1, 2);
if (!s) {
return;
}
p.Do(attackRate);
p.Do(decayRate);
p.Do(sustainRate);
p.Do(releaseRate);
p.Do(attackType);
p.Do(decayType);
p.Do(sustainType);
p.Do(sustainLevel);
p.Do(releaseType);
if (s < 2) {
p.Do(state_);
if (state_ == 4) {
state_ = STATE_OFF;
}
int stepsLegacy;
p.Do(stepsLegacy);
} else {
p.Do(state_);
}
p.Do(height_);
}